Displaying 20 results from an estimated 44 matches for "ipbx".
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ipb
2006 Dec 18
2
Digium TE405P with French E1 => Red Alert
...Clear channel (Default) (Slaves: 27)
Channel 28: Clear channel (Default) (Slaves: 28)
Channel 29: Clear channel (Default) (Slaves: 29)
Channel 30: Clear channel (Default) (Slaves: 30)
Channel 31: Clear channel (Default) (Slaves: 31)
31 channels configured.
but with all test, i have a red alert:
ipbx*CLI> zap show status
Description Alarms IRQ
bpviol CRC4
T4XXP (PCI) Card 0 Span 1 RED 0 0 0
T4XXP (PCI) Card 0 Span 2 UNCONFIGUR 0 0 0
T4XXP (PCI) Card 0 Span 3 UN...
2009 Nov 11
1
hosted / virtual IPBX platform
Hi,
I am looking for a hosted / virtual IPBX *PLATFORM* for service provider.Such hosted IPBX platform is aimed to be as a service, so that final customers don't have to install, maintain, or upgrade any PBX hardware or software.
It should have a control panel for end users to create / edit extension, conference rooms , IVR menus, etc
Cou...
2008 Dec 22
1
Web-driven SIP call thru Asterisk IPBX
Hi,
I think that the web-driven SIP Phone (free) doddle (beta version) can be useful with your Asterisk applications.
You can pre-fill it with your sip settings (Asterisk host name or IP /?realm / sip user), you just need to setup the HTML link as that: (Attached is the HTML page example)
?
/**************************/
simple HTML code example:
/*************************/
<html>
<head>
2006 Jan 27
0
pb with callerid
...UBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 25657 25657 IN IP4 10.101.51.252
s=session
c=IN IP4 10.101.51.252
t=0 0
m=audio 14324 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 7297
IPBX-TEST*CLI>
<-- SIP read from 10.101.51.248:2051:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport=5060
From: "DSOFT" <sip:123456789@10.101.51.252>;tag=as417ffda1
To: <sip:7297@10.101.51.248:2051;line=ld48ci1w>;tag=okikki5eaw
Call-ID: 6c...
2011 Sep 02
0
No subject
typing his number, though there is a 15 seconds timeout, and even if I type
the number very fast it still may happen to me.
*same => n,Read(mobileNumber,app/input-mobile,10,,2,15)*
In the logs:
When it fails:
- - <SIP/ipbx-iwred-000002e> Playing 'app/input-mobile.slin' (language 'fr')
- - User disconnected
When it succeeds:
- - <SIP/ipbx-iwred-000002e> Playing 'app/input-mobile.slin' (language 'fr')
- - User entered '0476123456'
The strange thing is that I cannot und...
2004 Jul 02
3
Termination for Asterisk Users - Inter-Asterisk Exchange
...any Asterisk PBX systems you have and can provide A-Z termination with immediate effect.
Any volume is good enough for us, even an amount as small as $1.00 a day will do for us.
We will provide connectivity from our Softswitch IP 216.162.116.46.
If anyone is interested, add this to your Asterisk IPBX and then email me for setting up a test account.
My email address is voip@netwebgroup.com
Thanks and have a great holiday weekend
Asha Kanuri
Netweb Group, Inc.
http://www.netwebgroup.com
2009 Jan 05
1
B410p, Ast1.4, France Télecom Numeris Double T0 problem
Hi.
When I call my RNIS numbers (with a mobile phone for example), I can
see 2 incoming calls on the IPBX, which should not happend.
I'm not sure if it's a problem with the telco France Telecom and their
ISDN setup, or if it's a problem
with the MISDN driver on the IPBX itself.
I'm stuck ...
Any advices for troubleshooting that?
Someone provide working configuration files for such s...
2006 Nov 04
0
iax2 qualify - false "peer unreachable"
...even when
two asterisks are in LAN environment,
please take a look into this debug, I can't find any problem with packet
loss, all qualify requests are replied and acknowledged,
I will submit bug report, if you will also not find any problems here...
PJ
BILL-GW:
bill*CLI> iax2 show peer ipbx-gw
Status : OK (4 ms)
Qualify : every 20000ms when OK, every 10000ms when UNREACHABLE
(sample smoothing On)
[Nov 4 11:06:58] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type:
IAX Subclass: POKE
[Nov 4 11:06:58] Timestamp: 00019ms SCall: 00002 DCall: 00000
[193.179....
2015 Jan 19
2
SEMI-OFFTOPIC openvox
Hi list, I write on the list looking for help, buy a openvox gw gsm
for four channels and I'm a little disappointed with the support
openvox, for some reason , The call doesn?t get trough
support tells me it was my asterisk server, but does not really work
me and my internal calls are working perfectly, I tested with another
sangoma FXO gateway and works perfectly.
the problem is that
2015 May 21
1
asterisk 13 webrtc
...encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=yes
transport=wss,ws
dtlsrekey=60
dtlsverify=no
dtlscertfile=/etc/pki/tls/certs/rapidssl.crt
dtlsprivatekey=/etc/pki/tls/private/rapidssl.key
dtlssetup=actpass
sip dump
<--- SIP read from WS:2.2.2.2:8558 --->
INVITE sip:887 at ipbx SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKyhVsmJdAtwRvuOWz9BHiNo1DGcqp4grQ;rport
From: "cervenka"<sip:vr1a882 at vhXXX.example.com>;tag=RDmpGm2Mubc5xQQ8NMli
To: <sip:887 at ipbx>
Contact:
"cervenka"<sips:vr1a882 at df7jal23ls0d.invalid;rtcweb-b...
2015 Jul 29
2
PJSIP T.38 issues
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Thanks for your reply Larry.
Le 27/07/2015 01:22, Larry Moore a ?crit :
> I think the "488 Not acceptable here" is occurring because the channel
> connecting through is not T.38 capable, that will be the IAX channel
> from iaxmomdem.
This is what T38gateway is supposed to do. And I'm very happy to report
that after one more
2004 Jan 16
1
ERROR[8192]
Hi all!
I get this error when trying to start asterisk:
ERROR[8192]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk
What can be the problem?
Thank you!
Miklos
iPFONE Telefonia IP
Rua Caio Graco 735 S?o Paulo SP
iPBX +55 11 3801-3702
UK +44 870 - 3403539
FWD 64662
sip:ipfone@sipserver.com.br
www.ipfone.com.br
info@ipfone.com.br
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2014 Apr 03
1
func_odbc
Hi All
Anyone know how to do include files with func_odbc.conf?
I now have several pages of functions in my func_odbc.conf and it is
getting harder to maintain it.
I would like to break them up into files by category. The standard method
of using the #include does not seem to work .
Ideas are appreciated.
Bryant
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2014 Jul 11
2
CDR(dst) not set in AEL macro
Hi
I'm using a macro to dial in a AEL dialplan. The problem is the macro do
not set the field CDR(dst), showing only ~~s~~.
I tried various configurations, but without solutions.
This is the macro:
macro dial-out(destno,dialstring,route_descr,interno) {
__TRANSFER_CONTEXT=ipbx;
if(${interno} = 1) {
Set(__PICKUPMARK=${destno});
if(${ODBC_verify_user(${CALLERID(num)})} > 0) {
t = tT;
} else {
t = t;
}
} else {
t = T;
}
Dial(${dialstring}/${destno},30,${t});
return;
}
Thank's.
Att,
*Rafael dos Santos Saraiva*
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/2...
2011 Mar 10
1
Is this true for Asterisk as SBC?
...a fully
functional Session Border Controller.
- IP phones can register with the SBC either from the internal network or
from the Internet.
- Use your SBC as an Inbound and/or Outbound proxy to have complete
control over incoming and outbound calls
- Use it to control access to your IPBX and to overcome the usual
problems associated with interfacing VoIP between your private network and
the Internet
- Solve one-way audio and other notoriously difficult and annoying NAT
traversal problems while, at the same time, improving your systems security
regards
dhaval
----------...
2008 Mar 21
1
Which command line is used to send emails to notify incoming voicemail ?
Hi,
In exim4, I can see lines such as :
mainlog.9:2008-03-12 08:53:28 1JZLmC-0000E7-0A <= root at foo.com U=root
P=local S=43802 id=Asterisk-0-123413860-4174-2662 at ipbx-bs-60200
In my voicemail.conf, I see :
; If you need to have an external program, i.e. /usr/bin/myapp called when a
;externnotify=/usr/bin/myapp
; If you need to have an external program, i.e. /usr/bin/myapp called when a
;externpass=/usr/bin/myapp
So, I guess this line (from app_voicemail.c) is...
2003 Aug 04
14
Mysql CDR
hello all,
I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record.
Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault.
the original version of cdr_mysql.so works fine but I need the start time and end
2003 Jun 28
0
SV: Newbie questions.....
...card to this? If so, could
> we then connect the Asterisk PBX to the callmanager?
> (Perhaps with another extension range).....and if so, how?
Yes, it's possible. I've done it to trombone calls through a
Operator system (Trio) and to interconnect with both Ericsson,
Nortel PBXs and iPBXs.
On CM you set it up as a trunk line, create a route map which
forward all calls to that specific E1 / T1 port on the 6608,
which are connected to the Asterisk Pri port.
On the Asterisk you do the same. Beware that top-down, bottom-up
is opposite on the 2 system. Then you should be able to get i...
2005 Aug 05
0
Seeking Beta testers for enterprise mystery service
...ling to experiment with new methods to make their VoIP (SIP)
platform more functional, secure, and cost-effective.
3) How much will it cost?
It's still too early to be asking that question. ;-) We expect
versions of the system to be affordable by any enterprise that is
implementing an iPBX-type platform, and there is almost certainly
going to be a free featureset that is generally available.
4) Why Asterisk?
Asterisk represents a very interesting community because:
a) Asterisk users tend to be more knowledgeable about VoIP
protocols and operations than users of vendor-speci...
2008 Jan 04
1
Remote hold on PRI
Hi everybody
We have a strange problem with several asterisk servers (Version
1.4.11) using PRI cards (tied to telco here in Belgium).
Indeed we noticed that whenever a local user places an outgoing call
through the PRI (and telco) to another IPBX (tied to telco using BRI
or PRI), if the remote party places the call on hold, the caller hears
the _local_ music on hold instead of the remote one. In fact we can
briefly hear the remote music on hold start, then it is replaced by
the local one.
More precisely:
Company 1 uses an asteris...